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But you are forgetting that sounds begin at some point in time. If you have 20 people banging on bottles with their house keys do you think 44.1kHz will capture all the "attacks" going on that the human ear can detect? How about 96kHz? From what I gather even 192kHz may not capture what the human ear can detect in terms of consecutive attacks. It's not all about reproducing a fourier transform of a bunch of continuous sine-waves......

Xenu wrote:All of you are banned, right?

Suppose the time interval between the two events is in the order of 22.68 microseconds - in that case they won't be encoded accurately with 44.1kHz sampling.

But, you may say, we can't hear anything above around 20kHz so big deal.... unfortunatly that only applies to continuous waveforms.

Beyond this, our host has pointed to his mastering of the Creedence albums for SACD vs. CD as an easy way to hear the difference generated by the difference in sampling rates and signal format (redbook vs. DSD). He claims, and others have backed him up with their observations, that the sense of ambience in at least one of the tracks is truncated by the lower sampling rate. By that I mean the sense of the spacial depth of reverberation as the sounds fade to silence, in other words, the signal decay envelope is reproduced differently and is clearly audible by anyone. Steve is certainly not the only one to make this type of observation, and you can easily verify this phenomenon for yourself if you doubt.

David, It seems that you are taking the view that the human ear is sensitive only to amplitude-response variation across a limited frequency range (20Hz -20 KHz). I would contend that human hearing also detects time-domain distortions or timing errors that cannot be measured directly in the frequency domain. Isn't that why jitter measurements are conducted nowadays (for one example) in addition to the standard frequency-response measurements when digital equipment is being designed and tested? Jitter has been demonstrated to be audible to very low thresholds. There are probably other time-domain distortions that are audible as well - perhaps quantization errors introduced by digital sampling?

This is why a 24/96 recorded 1kHz tone at 0dB sounds better than a 32 kbps MP3 1kHz tone at 0db. They'd both measure the same on a frequency analysis (barring any impure harmonics).

Xenu wrote:Because, of course, my deep dark secret is that unlike Krab and ThomH, I really don't specialize in this area, and being the sole person in that particular argument allows me to get bashed over the head with things like tri-delta sinosoidal waves that follow a fourier series, as I really don't know if that makes any sense. It's like evolution debates...I'm fine until it really gets down to the microbiological level.

I wish I had a quarter for each time I said it:
Your common sense makes you think of "more dots is better" or now "thin slices". But that "street's common sense" is leading you into the wrong conclusion. The slices need to be thin enough, but only to a point. This is Nyquist theory, and it is one of the main "legs of all modern technology. Say you want to have 20KHz audio. In theory, you need to take more then 40000 "slices" (40KHz sampling). In practice you may need to raise it a bit. The "reconnecting" ot the dots into a PERFECT wave (like th original analog one) is done by an analog filter. A perfect filter will make a perfect wave, at sample time and ALSO BETWEEN THE SAMPLES. That is why that "stuff" works.

The misunderstanding or lack of awarness by non technical people regarding Nyquist is the main reason behind that 192KHz hype. The electronics engineers understand Nyquist. The marketing types often do not. They took thier "street sense" into costly IC fabrication and "new wave of 192KHz gear", wrongly selling "more resolution". The sad thing - once they invested in that flawed concept, they refuse to take it back and acknowladge they simply took on what is over their heads...